1. Field of the Invention
The present invention relates generally to the field of telecommunications and more particularly to the field of transmitting aural information through a wide area network.
2. Description of Background Art
Conventional aural communication is accomplished using a public switched network, e.g., the telephone network. Through the use of such a public switched network users can easily communicate with each other by transmitting aural signals from a first terminal to a second terminal through the public switched network. The aural signals can represent voice data, modem data, or facsimile (fax) data, for example. In order to transmit voice data, a user performs a conventional procedure for setting up a call, and for tearing-down a call. In one example of a conventional call setup procedure, a user lifts a handset on a first telephone, listens for a dial-tone, and then enters a code identifying a destination telephone. A user at the destination telephone is notified that a connection is pending, e.g., by hearing the destination telephone ring, and the user lifts the handset. After the handset of the destination telephone is lifted, a connection between the first telephone and the destination telephone is established. As each user speaks, the sound is transformed and then transmitted through the telephone, through a private branch exchange (if any), through the public switched network, and then to the destination telephone where the transformed signal is re-transformed into an audible signal that can be heard by the user at the destination telephone.
Recently, communication systems have been developed that enable aural data to be transmitted over a wide area network (WAN). In these systems a private branch exchange (PBX) is connected to a communication device, e.g., a router, switch, FRAD, or multiplexor, that connects two networks having different data formats, e.g., a local area network (LAN) and a WAN. An example of a data format is a packet. A packet is a group of bits having a header portion and a data portion. The format of a packet can be different for each LAN and WAN. For example, the maximum size of a packet and packet destination and routing information can differ between networks. A router or a switch, hereafter referred to as a router, that also handles aural data converts a packet from a first format that is compatible with the PBX to a second format that is compatible with a WAN. After receiving the aural signals from the PBX, the router converts the aural signals into packets, transmits the packets across a WAN where they are received by a second router that is connected to a second PBX, key system or telephone. The second router converts the packets into aural signals and transmits the signals to the PBX, key system or telephone. There are problems with connecting a source of aural information directly to a router. One problem is that the format of aural information and the format of information that can be received by the network router are typically incompatible and, in general, a specially developed router must be used to enable the PBX to transmit data through a WAN. A second problem is that routers are, typically, not capable of being inexpensively modified to receive telephony functionality, for example, it is difficult to add a circuit board having the required telephony functionality to a router. Accordingly, in order to add telephony compatibility and functionality to a WAN, a WAN user must replace the existing routers. This is an expensive solution. A third problem is routers that are compatible with a PBX or a KTS generally provide proprietary solutions that are not compatible with those of other routers. A fourth problem is that such solutions are not generally available in routers, thus limiting the options of a user.
Another technique for transmitting aural signals across a WAN is to connect a microphone and a speaker to a conventional personal computer (PC). A user loads and executes a software program that converts the received analog signal to a digital signal using the processor in the PC. The signal may be sent over a LAN to a router. The router transmits the signal over a WAN to a second router. The second router may transmit the signal over a second LAN, or directly, to a destination PC. If the destination PC is operating compatible software, the PC can convert the received signal back to an audible signal that is transmitted through the PC's speakers. While this technique is less expensive than the first technique, it also has limitations. One limitation is that such systems are currently incapable of providing a priority mechanism that would ensure that aural data arrives within a predetermined maximum time period. Most data currently transmitted through WANs are not time sensitive, i.e., a small delay in receiving data is acceptable if the data is accurate. However aural communication is time sensitive, i.e., it is generally more important for aural data to be received in a timely manner than it is for the data to be absolutely accurate. If, while a user is speaking into a microphone, another computer that is coupled to a router via the LAN requests a transfer of a large file, e.g., a computer aided design (CAD) file, the packets of voice data that are received by the first router after the first router begins transmitting the packets of the CAD file may incur a significant delay if the router transmits all of the packets of the CAD file before transmitting the aural packets. In this situation, the second user will experience a significant delay in the reception of aural signals.
A second limitation is that due to limitations on host processing capability, the quality of the received aural signal is significantly degraded when compared to the transmitted aural signal and cannot be characterized as a toll-quality or near-toll-quality signal. The public switched networks in the industrialized countries provide a toll-quality signal. A near-toll-quality-signal is within 0.5 point of the toll-quality signal as measured by the means-opinion-score (MOS) method, on a scale of five, as determined by listening tests. A signal that is 1.0 point below the toll-quality signal is generally characterized as a communications-quality signal. The MOS method is described in greater detail in ITU-T Recommendation P.83, Subjective Performance Assessment of Telephone-Band and Wideband Digital CODECS, (March 1993), that is incorporated by reference herein in its entirety.
A third limitation is that the software for converting an analog aural signal from the microphone into an aural packet that is compatible with the LAN requires significant computational power that is provided by the processor in the PC. As a result of this additional computational load on the processor, the processing capability of the computer that is available for processing other application programs, is significantly reduced.
A fourth limitation is that the call setup and conversation between two users is not transparent. That is, the quality of the received signal is not at a near-toll-quality standard, and the procedures used to initiate the connection and to maintain the connection are not conventional. Instead of lifting a handset, listening for a dial-tone, and entering a destination identifier on a telephone keypad, the user executes a software program, types a destination identifier using a computer keyboard and then talks into an external microphone and listens though speakers attached to the PC. Similarly, to receive the audible signals, the user must, for example, execute a software program, use a mouse to click on an "answer" button, and turn on the speakers. This lack of transparency requires users to re-learn how to communicate with a person at a remote location.
Accordingly, what is needed is a system and method: (1) for transmitting aural information as digital signals over a wide-area-data-network; (2) for transparently generating and receiving aural data; (3) that incorporates a robust error correction procedure that enables a receiver to recreate lost data; (4) that converts aural signals into a network compatible format, and that performs compression and decompression algorithms on the converted data without placing a significant computational load on a host processor; (5) that uses a router/switch priority system to minimize the end-to-end packet delay across a wide area network; (6) that adjusts the destination signal based upon packet delay variations; (7) that communicates with a router/switch over a standard LAN connection without requiring a specialized router/switch voice interface; and (8) that can connect to a LAN with a standard interface and can communicate over the LAN in standard data formats.